I am just starting with Asterisk .. I think you are right, I am doing an attended transfer, although I don't exactly understand what that means. I still need to know in what lot I can pickup my call again right?
Ok, my config .. (i will leave out the commented stuff, because there's lot of comments in the sample config) [general] parkext => 700 ; What extension to dial to park. Set per parking lot. parkpos => 701-720 ; What extensions to park calls on. (defafult parking lot) context => parkedcalls ; Which context parked calls are in (default parking lot) parkingtime => 300 ; Number of seconds a call can be parked before returning. comebacktoorigin = yes ; Setting this option configures the behavior of call parking when the courtesytone = beep ; Sound file to play to when someone picks up a parked call parkedplay = both ; Who to play courtesytone to when picking up a parked call. Thanks! On Mon, Jan 16, 2012 at 4:59 PM, Eric Wieling <[email protected]> wrote: > This symptom usually means you are doing an attended transfer instead of a > blind transfer. > > -----Original Message----- > From: [email protected] [mailto: > [email protected]] On Behalf Of Roland > Sent: Monday, January 16, 2012 10:57 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] SayDigits playback doesn't always work > > Ok, got it. Indeed, starting with Answer() helped. > > But I still don't understand why the parking feature isn't working then. I > used the sample config. Transfer the call to 700, playback of the lot is > being executed, but I hear nothing. Probably the same problem, but how do I > change this? > > This is the call that doesn't work. Then when I call 200, I see > this: > > > > [Jan 16 15:54:29] == Using SIP RTP CoS mark 5 > > [Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new > state InUse for Notify User 001565150F04.1 > > [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1] > Answer("SIP/000B822FD265-0000003e", "") in new stack > > [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2] > BackGround("SIP/000B822FD265-0000003e", "main-menu") in new stack > > [Jan 16 15:54:29] -- <SIP/000B822FD265-0000003e> Playing > 'main-menu.gsm' (language 'nl') > > [Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3] > WaitExten("SIP/000B822FD265-0000003e", "5") in new stack > > [Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-0000003e > > [Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1] > Wait("SIP/000B822FD265-0000003e", "2") in new stack > > [Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2] > SayDigits("SIP/000B822FD265-0000003e", "123") in new stack > > [Jan 16 15:54:36] -- <SIP/000B822FD265-0000003e> Playing > 'digits/1.gsm' (language 'nl') > > [Jan 16 15:54:36] -- <SIP/000B822FD265-0000003e> Playing > 'digits/2.gsm' (language 'nl') > > [Jan 16 15:54:37] -- <SIP/000B822FD265-0000003e> Playing > 'digits/3.gsm' (language 'nl') > > [Jan 16 15:54:37] -- Auto fallthrough, channel > 'SIP/000B822FD265-0000003e' status is 'UNKNOWN' > > [Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new > state Idle for Notify User 001565150F04.1 > > > > This call works perfectly. What am I missing? > > > > In my sip.conf I have: > > > > [stumpel-zwaag](!) ; create template > for our devices > > type=friend ; the channel > driver will mathc on username first, IP second > > context=StumpelZwaag ; this is where > calls from the device will enter the dialplan > > host=dynamic ; the device will > register with asterisk > > ;nat=yes ; assume > the device is behind nat > > secret=xxx ; a secure password for > this device > > dtmfmode=auto ; accept > touch-tones from devices, negotiated automatically > > disallow=all ; reset with voice > codecs to accept from, and request to, the device > > allow=alaw ; which audio > codecs we accept from > > canreinvite=nonat > > > > > > > -- > > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com-- > New to Asterisk? Join us for a live introductory webinar every > Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
